mirror of
https://github.com/nyanmisaka/ffmpeg-rockchip.git
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Merge commit '40aaa8dadfd1c69ff4460d04750e1403b5535a6d'
* commit '40aaa8dadfd1c69ff4460d04750e1403b5535a6d': examples/avcodec: split audio encoding into a separate example Merged-by: Clément Bœsch <u@pkh.me>
This commit is contained in:
@@ -44,197 +44,6 @@
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#define AUDIO_INBUF_SIZE 20480
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#define AUDIO_REFILL_THRESH 4096
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/* check that a given sample format is supported by the encoder */
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static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt)
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{
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const enum AVSampleFormat *p = codec->sample_fmts;
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while (*p != AV_SAMPLE_FMT_NONE) {
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if (*p == sample_fmt)
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return 1;
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p++;
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}
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return 0;
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}
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/* just pick the highest supported samplerate */
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static int select_sample_rate(AVCodec *codec)
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{
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const int *p;
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int best_samplerate = 0;
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if (!codec->supported_samplerates)
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return 44100;
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p = codec->supported_samplerates;
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while (*p) {
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best_samplerate = FFMAX(*p, best_samplerate);
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p++;
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}
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return best_samplerate;
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}
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/* select layout with the highest channel count */
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static int select_channel_layout(AVCodec *codec)
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{
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const uint64_t *p;
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uint64_t best_ch_layout = 0;
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int best_nb_channels = 0;
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if (!codec->channel_layouts)
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return AV_CH_LAYOUT_STEREO;
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p = codec->channel_layouts;
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while (*p) {
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int nb_channels = av_get_channel_layout_nb_channels(*p);
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if (nb_channels > best_nb_channels) {
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best_ch_layout = *p;
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best_nb_channels = nb_channels;
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}
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p++;
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}
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return best_ch_layout;
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}
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/*
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* Audio encoding example
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*/
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static void audio_encode_example(const char *filename)
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{
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AVCodec *codec;
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AVCodecContext *c= NULL;
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AVFrame *frame;
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AVPacket pkt;
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int i, j, k, ret, got_output;
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int buffer_size;
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FILE *f;
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uint16_t *samples;
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float t, tincr;
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printf("Encode audio file %s\n", filename);
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/* find the MP2 encoder */
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codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
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if (!codec) {
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fprintf(stderr, "Codec not found\n");
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exit(1);
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}
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c = avcodec_alloc_context3(codec);
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if (!c) {
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fprintf(stderr, "Could not allocate audio codec context\n");
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exit(1);
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}
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/* put sample parameters */
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c->bit_rate = 64000;
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/* check that the encoder supports s16 pcm input */
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c->sample_fmt = AV_SAMPLE_FMT_S16;
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if (!check_sample_fmt(codec, c->sample_fmt)) {
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fprintf(stderr, "Encoder does not support sample format %s",
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av_get_sample_fmt_name(c->sample_fmt));
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exit(1);
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}
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/* select other audio parameters supported by the encoder */
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c->sample_rate = select_sample_rate(codec);
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c->channel_layout = select_channel_layout(codec);
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c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
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/* open it */
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if (avcodec_open2(c, codec, NULL) < 0) {
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fprintf(stderr, "Could not open codec\n");
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exit(1);
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}
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f = fopen(filename, "wb");
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if (!f) {
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fprintf(stderr, "Could not open %s\n", filename);
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exit(1);
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}
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/* frame containing input raw audio */
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frame = av_frame_alloc();
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if (!frame) {
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fprintf(stderr, "Could not allocate audio frame\n");
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exit(1);
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}
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frame->nb_samples = c->frame_size;
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frame->format = c->sample_fmt;
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frame->channel_layout = c->channel_layout;
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/* the codec gives us the frame size, in samples,
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* we calculate the size of the samples buffer in bytes */
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buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size,
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c->sample_fmt, 0);
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if (buffer_size < 0) {
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fprintf(stderr, "Could not get sample buffer size\n");
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exit(1);
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}
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samples = av_malloc(buffer_size);
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if (!samples) {
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fprintf(stderr, "Could not allocate %d bytes for samples buffer\n",
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buffer_size);
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exit(1);
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}
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/* setup the data pointers in the AVFrame */
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ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
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(const uint8_t*)samples, buffer_size, 0);
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if (ret < 0) {
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fprintf(stderr, "Could not setup audio frame\n");
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exit(1);
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}
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/* encode a single tone sound */
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t = 0;
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tincr = 2 * M_PI * 440.0 / c->sample_rate;
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for (i = 0; i < 200; i++) {
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av_init_packet(&pkt);
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pkt.data = NULL; // packet data will be allocated by the encoder
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pkt.size = 0;
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for (j = 0; j < c->frame_size; j++) {
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samples[2*j] = (int)(sin(t) * 10000);
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for (k = 1; k < c->channels; k++)
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samples[2*j + k] = samples[2*j];
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t += tincr;
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}
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/* encode the samples */
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ret = avcodec_encode_audio2(c, &pkt, frame, &got_output);
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if (ret < 0) {
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fprintf(stderr, "Error encoding audio frame\n");
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exit(1);
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}
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if (got_output) {
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fwrite(pkt.data, 1, pkt.size, f);
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av_packet_unref(&pkt);
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}
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}
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/* get the delayed frames */
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for (got_output = 1; got_output; i++) {
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ret = avcodec_encode_audio2(c, &pkt, NULL, &got_output);
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if (ret < 0) {
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fprintf(stderr, "Error encoding frame\n");
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exit(1);
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}
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if (got_output) {
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fwrite(pkt.data, 1, pkt.size, f);
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av_packet_unref(&pkt);
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}
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}
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fclose(f);
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av_freep(&samples);
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av_frame_free(&frame);
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avcodec_free_context(&c);
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}
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/*
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* Audio decoding.
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*/
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@@ -646,7 +455,6 @@ int main(int argc, char **argv)
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if (!strcmp(output_type, "h264")) {
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video_encode_example("test.h264", AV_CODEC_ID_H264);
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} else if (!strcmp(output_type, "mp2")) {
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audio_encode_example("test.mp2");
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audio_decode_example("test.pcm", "test.mp2");
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} else if (!strcmp(output_type, "mpg")) {
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video_encode_example("test.mpg", AV_CODEC_ID_MPEG1VIDEO);
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